audio_encoder_copy_red.h

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/*

 *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.

 *  Use of this source code is governed by a BSD-style license

 *  that can be found in the LICENSE file in the root of the source

 *  tree. An additional intellectual property rights grant can be found

 *  in the file PATENTS.  All contributing project authors may

 *  be found in the AUTHORS file in the root of the source tree.

*/

#ifndef MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_

#define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_

#include <stddef.h>

#include <stdint.h>

#include <list>

#include <memory>

#include <utility>

#include "absl/types/optional.h"

#include "api/array_view.h"

#include "api/audio_codecs/audio_encoder.h"

#include "api/field_trials_view.h"

#include "api/units/time_delta.h"

#include "rtc_base/buffer.h"

namespace webrtc {

// This class implements redundant audio coding as described in

//   https://tools.ietf.org/html/rfc2198

// The class object will have an underlying AudioEncoder object that performs

// the actual encodings. The current class will gather the N latest encodings

// from the underlying codec into one packet. Currently N is hard-coded to 2.

class AudioEncoderCopyRed final : public AudioEncoder {

 public:

  struct Config {

    Config();

    Config(Config&&);

    ~Config();

    int payload_type;

    std::unique_ptr<AudioEncoder> speech_encoder;

};

  AudioEncoderCopyRed(Config&& config, const FieldTrialsView& field_trials);

  ~AudioEncoderCopyRed() override;

  AudioEncoderCopyRed(const AudioEncoderCopyRed&) = delete;

  AudioEncoderCopyRed& operator=(const AudioEncoderCopyRed&) = delete;

  int SampleRateHz() const override;

  size_t NumChannels() const override;

  int RtpTimestampRateHz() const override;

  size_t Num10MsFramesInNextPacket() const override;

  size_t Max10MsFramesInAPacket() const override;

  int GetTargetBitrate() const override;

  void Reset() override;

  bool SetFec(bool enable) override;

  bool SetDtx(bool enable) override;

  bool GetDtx() const override;

  bool SetApplication(Application application) override;

  void SetMaxPlaybackRate(int frequency_hz) override;

  bool EnableAudioNetworkAdaptor(const std::string& config_string,

                                 RtcEventLog* event_log) override;

  void DisableAudioNetworkAdaptor() override;

  void OnReceivedUplinkPacketLossFraction(

      float uplink_packet_loss_fraction) override;

  void OnReceivedUplinkBandwidth(

      int target_audio_bitrate_bps,

      absl::optional<int64_t> bwe_period_ms) override;

  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;

  void OnReceivedRtt(int rtt_ms) override;

  void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;

  void SetReceiverFrameLengthRange(int min_frame_length_ms,

                                   int max_frame_length_ms) override;

  ANAStats GetANAStats() const override;

  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()

      const override;

  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()

      override;

 protected:

  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,

                         rtc::ArrayView<const int16_t> audio,

                         rtc::Buffer* encoded) override;

 private:

  std::unique_ptr<AudioEncoder> speech_encoder_;

  rtc::Buffer primary_encoded_;

  size_t max_packet_length_;

  int red_payload_type_;

  std::list<std::pair<EncodedInfo, rtc::Buffer>> redundant_encodings_;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_